search for: 20msec

...ticks correctly but I think it can be set to 20(msec). It is set as a Member in my class and i pointed out where i use it so it is clear if i use it somewhere i better shouldnt. - I initialize the JitterBuffer with ticks = 20 (saved in the variable m_Ticks) - My network thread repeadedly (every 20msec) calls AddPacket() which adds the packet to the buffer 320 bytes of audio data are set as the data the timestamp is set to the sequence number of the packet times 20 (m_Ticks) the span of the packet is set to 20 (the packet covers one tick entirely) - A thread which...

...ec). >> It is set as a Member in my class and i pointed out where i use it so it >> is clear if i use it somewhere i better shouldnt. >> >> - I initialize the JitterBuffer with ticks = 20 (saved in the variable >> m_Ticks) >> - My network thread repeadedly (every 20msec) calls AddPacket() which >> adds the packet to the buffer >> 320 bytes of audio data are set as the data >> the timestamp is set to the sequence number of the packet times >> 20 (m_Ticks) >> the span of the packet is set to 20 (the packet cov...

...(12 byes Header) tells about the version, payload time, time stamp, srsc, etc. The rest of the packet is OPUS payload (raw format), The TOC byte from OPUS payload tells its 20ms frame, even the time stamp different of 960 means 20 msec frame. Questions: 1) does 960 means number of samples in the (20msec frame)? (fs=48KHz) 2) The raw opus packet (i.e OPUS payload), how can I convert it to an playable format? 3) is there any information OPUS payload has(except TOC(first byte))??? 4) I have RAW opus packets now, I want to convert it to representable files, integer, float any method?? or any help (dec...

Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_AGC,...

Hi, I looked at the jitter buffer code and it seems like the maximum number of frames that Speex can hold up to is 200 (SPEEX_JITTER_MAX_BUFFER_SIZE). This is equivalent to 4 seconds (20msec/frame * 200 frames). Can I just reduce this constant to limit the size? I know I'll reduce the smoothness of delayed frames but I want to reduce the delay in case my audio application isn't servicing the frames in time and hence the delay will grow as more frames are queued up. I pretty...

...an be set to > 20(msec). > It is set as a Member in my class and i pointed out where i use it so it > is clear if i use it somewhere i better shouldnt. > > - I initialize the JitterBuffer with ticks = 20 (saved in the variable > m_Ticks) > - My network thread repeadedly (every 20msec) calls AddPacket() which > adds the packet to the buffer > 320 bytes of audio data are set as the data > the timestamp is set to the sequence number of the packet times > 20 (m_Ticks) > the span of the packet is set to 20 (the packet covers one tick > en...

...buffer? The way my code works today is: 1) Packet read from socket 2) Call speex_jitter_put(...) with the just-arrived packet 3) Read one packet from jitter buffer using speex_jitter_get(...) function 4) Feed just read-from-jitter packet to the sound card for playback This will in fact feed one 20msec batch of sound to play at the sound card for every packet received from the speex encoder at the other end. I know I may sound a bit slow-on-the-pickup here, but at the risk of sounding very beginner like (which I'll gladly admit I am) I wonder if this is totally wrong to do? Question: Should...

...set as a Member in my class and i pointed out where i use it so it >>> is clear if i use it somewhere i better shouldnt. >>> >>> - I initialize the JitterBuffer with ticks = 20 (saved in the variable >>> m_Ticks) >>> - My network thread repeadedly (every 20msec) calls AddPacket() which >>> adds the packet to the buffer >>> 320 bytes of audio data are set as the data >>> the timestamp is set to the sequence number of the packet times >>> 20 (m_Ticks) >>> the span of the packet is set t...

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are generated: Apr 20 17:37:27 NOTICE[20...

...oday is: > > 1) Packet read from socket > 2) Call speex_jitter_put(...) with the just-arrived packet > 3) Read one packet from jitter buffer using speex_jitter_get(...) function > 4) Feed just read-from-jitter packet to the sound card for playback > > This will in fact feed one 20msec batch of sound to play at the sound card > for every packet received from the speex encoder at the other end. > > I know I may sound a bit slow-on-the-pickup here, but at the risk of > sounding very beginner like (which I'll gladly admit I am) I wonder if this > is totally wrong...

...> to be within [0, 1) in Q15 format. Then based on the speech energy levels, > in Step2, the probability is updated whose value shall also lie between [0, > 1). Later in Step3, the smooth coeff is calculated. This code do not have > any issue when the frame size is more than or equal to 20msec. But, if the > frame size is 10ms, then in step2, the energy is doubled (this may be done > because the original Silk code is for 20ms. To convert the energy for 20ms, > it could have been doubled). When this is done the probability which is > updated in step2 becomes more than 1. When...

Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); } /* Modifica...

...oday is: > > 1) Packet read from socket > 2) Call speex_jitter_put(...) with the just-arrived packet > 3) Read one packet from jitter buffer using speex_jitter_get(...) function > 4) Feed just read-from-jitter packet to the sound card for playback > > This will in fact feed one 20msec batch of sound to play at the sound card > for every packet received from the speex encoder at the other end. > > I know I may sound a bit slow-on-the-pickup here, but at the risk of > sounding very beginner like (which I'll gladly admit I am) I wonder if this > is totally wrong...

...e value is expected to be within [0, 1) in Q15 format. Then based on the speech energy levels, in Step2, the probability is updated whose value shall also lie between [0, 1). Later in Step3, the smooth coeff is calculated. This code do not have any issue when the frame size is more than or equal to 20msec. But, if the frame size is 10ms, then in step2, the energy is doubled (this may be done because the original Silk code is for 20ms. To convert the energy for 20ms, it could have been doubled). When this is done the probability which is updated in step2 becomes more than 1. When this is used in mult...

On 2016-06-17 10:50 AM, Rizwan Ishaq wrote: > 1) does 960 means number of samples in the (20msec frame)? (fs=48KHz) That's correct. > 2) The raw opus packet (i.e OPUS payload), how can I convert it to an > playable format? You can pass the payload section of the RTP packets directly to an opus decoder to convert to PCM float or integer samples. To save it in a file other audio pl...

Hi All, I have taken ITU standard 48KHz female voice file and Encoded using opus_demo by giving follwoing arguments. opus_demo -e audio 48000 1 32000 female1.pcm female1frame.opus And using repacketiser tried to pack 6frames/packet(6*20msec=120msec). While packing I observed below error for 3 times. Error: opus_repacketizer_cat() failed: corrupted stream when i inspect the encoded file, it looks repacketiser fails to pack a encoded frames of different modes( HYBRID, CELT) in a single packet. 1. Does above mentioned issue is the curr...

...e value is expected to be within [0, 1) in Q15 format. Then based on the speech energy levels, in Step2, the probability is updated whose value shall also lie between [0, 1). Later in Step3, the smooth coeff is calculated. This code do not have any issue when the frame size is more than or equal to 20msec. But, if the frame size is 10ms, then in step2, the energy is doubled (this may be done because the original Silk code is for 20ms. To convert the energy for 20ms, it could have been doubled). When this is done the probability which is updated in step2 becomes more than 1. When this is used in mult...

Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts

...the speech energy levels, in Step2, the probability is > updated whose value shall also lie between [0, 1). Later in > Step3, the smooth coeff is calculated. This code do not have > any issue when the frame size is more than or equal to > 20msec. But, if the frame size is 10ms, then in step2, the > energy is doubled (this may be done because the original > Silk code is for 20ms. To convert the energy for 20ms, it > could have been doubled). When this is done the probability > whic...

...ithin [0, 1) in Q15 format. Then based on the speech energy levels, >> in Step2, the probability is updated whose value shall also lie between [0, >> 1). Later in Step3, the smooth coeff is calculated. This code do not have >> any issue when the frame size is more than or equal to 20msec. But, if the >> frame size is 10ms, then in step2, the energy is doubled (this may be done >> because the original Silk code is for 20ms. To convert the energy for 20ms, >> it could have been doubled). When this is done the probability which is >> updated in step2 becomes mo...